通过webRTC+websocket实现视频通话

最近空闲时间较多,于是写了一个个人博客,其中实现了实时通话,但是感觉功能比较无趣,前两天突发奇想,觉得如果可以视频通话就有意思了,于是从网上搜索了相关资料,发现webRTC可以实现该功能,但是网上关于webRTC的资料甚少,经过三天的研究,终于简单的实现了视频通话的功能,具体实现过程如下:

1.技术点:

webRTC、websocket

2.实现思路:

两个浏览器打开同一页面,连接到同一个socket。
此时由一端点击建立连接,发起建立连接的一端就是offer(携带信号源信息),发给另外一个端,另外一个端收到offer之后,发出响应answer(携带信号源信息),offer端收到answer端信息进行存储;这样每个端都有了自己的信息和对方的信息,offer发出answer发出后设置了localDescription和remoteDescription后就会触发onicecandidate,如此一来,双方都有了对方的localDescription、remoteDescription和candidata;有了这三种数据之后,就可以触发Connection.onaddstream函数,然后通过theirVideo.srcObject = e.stream这个方法,把流写到video标签内,然后video标签里就会有对方的视频画面了。

注意:这样实现之后不能直接调用navigator.getUserMedia()函数,浏览器会默认getUserMedia为undifined,如果是互联网模式这里需要设置浏览(如果只是本地测试则不需要)(谷歌浏览器配置方法);webrtc如只是p2p不需要特别服务器,自已开发信令服务就可以啦,当要安装turn server 国内常有打洞不成功需要转发。

下面是实现代码:

1.前端页面

<!DOCTYPE html>
<html lang="en">
<head>
    <meta charset="UTF-8">
    <meta name="viewport" content="width=device-width, initial-scale=1.0">
    <meta http-equiv="X-UA-Compatible" content="ie=edge">
    <title>webrtc</title>
    <style>
        #yours{
            width:300px;
            position:absolute;
            top:200px;
            left:100px;
        }
        #theirs{
            width:300px;
            position:absolute;
            top:200px;
            left:400px;
        }
    </style>
</head>
<body>
<button onclick="createOffer()">开始视频</button>
<video id="yours" autoplay></video>
<video id="theirs" autoplay></video>

</body>

<script type="text/javascript" src="../blog/plugin/jquery/1.9.1/jquery.min.js"></script>
<script src="webtrc.js"></script>
<script>
    $(function () {
        f()
    })
</script>
</html>

2.js、webRTC实现方法

var websocket;

function randomNum(minNum,maxNum){
    switch(arguments.length){
        case 1:
            return parseInt(Math.random()*minNum+1,10);
            break;
        case 2:
            return parseInt(Math.random()*(maxNum-minNum+1)+minNum,10);
            break;
        default:
            return 0;
            break;
    }
}
const userid = 'user' + randomNum(0,100000);

function hasUserMedia() {
    navigator.getUserMedia = navigator.getUserMedia || navigator.msGetUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
    return !!navigator.getUserMedia;
}
function hasRTCPeerConnection() {
    window.RTCPeerConnection = window.RTCPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection || window.msRTCPeerConnection;
    return !!window.RTCPeerConnection;
}

var yourVideo = document.getElementById("yours");
var theirVideo = document.getElementById("theirs");
var Connection;


function startPeerConnection() {
    //return;
    var config = {
        'iceServers': [
            //{ 'urls': 'stun:stun.xten.com:3478' },
            //{ 'urls': 'stun:stun.voxgratia.org:3478' },
            { 'url': 'stun:stun.l.google.com:19302' }
        ]
    };
    config = {
        iceServers: [
            { urls: 'stun:stun.l.google.com:19302' },
            { urls: 'stun:global.stun.twilio.com:3478?transport=udp' }
        ],
        //sdpSemantics: 'unified-plan'
    };
    // {
    //     "iceServers": [{
    //         "url": "stun:stun.1.google.com:19302"
    //     }]
    // };
    Connection = new RTCPeerConnection(config);
    Connection.onicecandidate = function(e) {
        console.log('onicecandidate');
        if (e.candidate) {
            websocket.send(JSON.stringify({
                "userid":userid,
                "event": "_ice_candidate",
                "data": {
                    "candidate": e.candidate
                }
            }));
        }
    }
    Connection.onaddstream = function(e) {
        console.log('onaddstream');

        //theirVideo.src = window.URL.createObjectURL(e.stream);
        theirVideo.srcObject = e.stream;
    }
}


createSocket();
startPeerConnection();

function f() {
    navigator.getUserMedia({ video: true, audio: true },
        stream => {
            yourVideo.srcObject = stream;
            window.stream = stream;
            Connection.addStream(stream)
        },
        err => {
            console.log(err);
        })
}


function createOffer(){
    //发送offer和answer的函数,发送本地session描述
    Connection.createOffer().then(offer => {
        Connection.setLocalDescription(offer);
        websocket.send(JSON.stringify({
            "userid":userid,
            "event": "offer",
            "data": {
                "sdp": offer
            }
        }));
    });
}



function createSocket(){
    var user = Math.round(Math.random()*1000) + ""
    websocket = new WebSocket("ws://127.0.0.1:8080/msgServer/"+user);
    eventBind();
};
function eventBind() {
    //连接成功
    websocket.onopen = function(e) {
        console.log('连接成功')
    };
    //server端请求关闭
    websocket.onclose = function(e) {
        console.log('close')
    };
    //error
    websocket.onerror = function(e) {

    };
    //收到消息
    websocket.onmessage = (event)=> {
        if(event.data == "new user") {
            location.reload();
        } else {
            var json = JSON.parse(event.data);
            console.log('onmessage: ', json);
            if(json.userid !=userid){
                //如果是一个ICE的候选,则将其加入到PeerConnection中,否则设定对方的session描述为传递过来的描述
                if(json.event === "_ice_candidate"&&json.data.candidate) {
                    Connection.addIceCandidate(new RTCIceCandidate(json.data.candidate));
                }else if(json.event ==='offer'){
                    Connection.setRemoteDescription(json.data.sdp);
                    Connection.createAnswer().then(answer => {
                        Connection.setLocalDescription(answer);
                        console.log(window.stream)
                        websocket.send(JSON.stringify({
                            "userid":userid,
                            "event": "answer",
                            "data": {
                                "sdp": answer
                            }
                        }));
                    })
                }else if(json.event ==='answer'){
                    Connection.setRemoteDescription(json.data.sdp);
                    console.log(window.stream)
                }
            }
        }
    };
}

websocket

package com.chat.api.web.restctrl;

import org.springframework.context.annotation.Scope;
import org.springframework.stereotype.Component;
import org.springframework.util.StringUtils;

import javax.websocket.*;
import javax.websocket.server.PathParam;
import javax.websocket.server.ServerEndpoint;
import java.io.IOException;
import java.util.Enumeration;
import java.util.concurrent.ConcurrentHashMap;


@ServerEndpoint("/msgServer/{userId}")
@Component
@Scope("prototype")
public class WebSocketServer {

    /**
     * 静态变量,用来记录当前在线连接数。应该把它设计成线程安全的。
     */
    private static int onlineCount = 0;
    /**
     * concurrent包的线程安全Set,用来存放每个客户端对应的MyWebSocket对象。
     */
    private static ConcurrentHashMap<String, Session> webSocketMap = new ConcurrentHashMap<>();
    /**
     * 与某个客户端的连接会话,需要通过它来给客户端发送数据
     */
    private Session session;
    /**
     * 接收userId
     */
    private String userId = "";

    @OnOpen
    public void onOpen(Session session, @PathParam("userId") String userId) {
        this.session = session;
        this.userId = userId;
        /**
         * 连接被打开:向socket-map中添加session
         */
        webSocketMap.put(userId, session);
        System.out.println(userId + " - 连接建立成功...");
    }

    @OnMessage
    public void onMessage(String message, Session session) {
        try {
            this.sendMessage(message);
        } catch (IOException e) {
            e.printStackTrace();
        }
    }

    @OnError
    public void onError(Session session, Throwable error) {
        System.out.println("连接异常...");
        error.printStackTrace();
    }

    @OnClose
    public void onClose() {
        System.out.println("连接关闭");
    }

    /**
     * 实现服务器主动推送
     */
    public void sendMessage(String message) throws IOException {
        if (message.equals("心跳")){
            this.session.getBasicRemote().sendText(message);
        }
        Enumeration<String> keys = webSocketMap.keys();
        while (keys.hasMoreElements()){
            String key = keys.nextElement();
            if (key.equals(this.userId)){
                System.err.println("my id " + key);
                continue;
            }
            if (webSocketMap.get(key) == null){
                webSocketMap.remove(key);
                System.err.println(key + " : null");
                continue;
            }
            Session sessionValue = webSocketMap.get(key);
            if (sessionValue.isOpen()){
                System.out.println("发消息给: " + key + " ,message: " + message);
                sessionValue.getBasicRemote().sendText(message);
            }else {
                System.err.println(key + ": not open");
                sessionValue.close();
                webSocketMap.remove(key);
            }
        }
    }

    /**
     * 发送自定义消息
     */
    public static void sendInfo(String message, @PathParam("userId") String userId) throws IOException {
        System.out.println("发送消息到:" + userId + ",内容:" + message);
        if (!StringUtils.isEmpty(userId) && webSocketMap.containsKey(userId)) {
            webSocketMap.get(userId).getBasicRemote().sendText(message);
            //webSocketServer.sendMessage(message);
        } else {
            System.out.println("用户" + userId + ",不在线!");
        }
    }

    public static synchronized int getOnlineCount() {
        return onlineCount;
    }

    public static synchronized void addOnlineCount() {
        WebSocketServer.onlineCount++;
    }

    public static synchronized void subOnlineCount() {
        WebSocketServer.onlineCount--;
    }
}

最终的实现效果如下(为了避免我的帅气伤人,这里打码( ̄ェ ̄;))

在这里插入图片描述

自己尝试的搭建了信令服务器,但是没有成功,已经购买了域名,备案成功后,想通过域名来实现跨网的视频通话。

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实现局域网音视频通话可以用Spring Boot作为后端框架,Netty作为网络通信框架,WebSocket作为实现双向通信的协议。以下是一个简单的实现过程: 1. 首先需要搭建一个Spring Boot项目,可以使用Spring Initializr来快速生成项目。在pom.xml中添加Netty和WebSocket的依赖,例如: ```xml <dependency> <groupId>io.netty</groupId> <artifactId>netty-all</artifactId> <version>4.1.25.Final</version> </dependency> <dependency> <groupId>org.springframework.boot</groupId> <artifactId>spring-boot-starter-websocket</artifactId> </dependency> ``` 2. 创建一个WebSocket处理器类,用来处理WebSocket的连接、关闭和消息收发等逻辑。例如: ```java @Component @ServerEndpoint("/video-chat") public class VideoChatHandler { private static final Logger LOGGER = LoggerFactory.getLogger(VideoChatHandler.class); @OnOpen public void onOpen(Session session) { LOGGER.info("WebSocket opened: {}", session.getId()); } @OnMessage public void onMessage(String message, Session session) { LOGGER.info("Received message: {}", message); // TODO: 处理收到的消息 } @OnClose public void onClose(Session session) { LOGGER.info("WebSocket closed: {}", session.getId()); } @OnError public void onError(Throwable error) { LOGGER.error("WebSocket error", error); } } ``` 3. 在Spring Boot的配置类中添加WebSocket的配置,例如: ```java @Configuration @EnableWebSocket public class WebSocketConfig implements WebSocketConfigurer { @Autowired private VideoChatHandler videoChatHandler; @Override public void registerWebSocketHandlers(WebSocketHandlerRegistry registry) { registry.addHandler(videoChatHandler, "/video-chat").setAllowedOrigins("*"); } } ``` 4. 使用Netty来实现视频的传输。可以使用Netty提供的UDP协议来实现多人音视频通话,也可以使用TCP协议来实现点对点的音视频通话。需要根据实际情况选择相应的协议,这里以TCP协议为例: ```java @Component public class VideoChatServer { private static final Logger LOGGER = LoggerFactory.getLogger(VideoChatServer.class); @Value("${server.video-chat.port}") private int port; @PostConstruct public void start() { EventLoopGroup bossGroup = new NioEventLoopGroup(); EventLoopGroup workerGroup = new NioEventLoopGroup(); try { ServerBootstrap bootstrap = new ServerBootstrap(); bootstrap.group(bossGroup, workerGroup) .channel(NioServerSocketChannel.class) .childHandler(new ChannelInitializer<SocketChannel>() { @Override public void initChannel(SocketChannel ch) throws Exception { ChannelPipeline pipeline = ch.pipeline(); // TODO: 添加音视频相关的编解码器和处理器 } }) .option(ChannelOption.SO_BACKLOG, 128) .childOption(ChannelOption.SO_KEEPALIVE, true); ChannelFuture future = bootstrap.bind(port).sync(); LOGGER.info("Video chat server started on port {}", port); future.channel().closeFuture().sync(); } catch (InterruptedException e) { LOGGER.error("Video chat server interrupted", e); } finally { workerGroup.shutdownGracefully(); bossGroup.shutdownGracefully(); } } } ``` 5. 在WebSocket处理器中实现视频数据的收发逻辑。当收到音视频数据时,可以将数据转发给所有连接的WebSocket客户端。例如: ```java @Component @ServerEndpoint("/video-chat") public class VideoChatHandler { private static final Logger LOGGER = LoggerFactory.getLogger(VideoChatHandler.class); private List<Session> sessions = new CopyOnWriteArrayList<>(); @OnOpen public void onOpen(Session session) { LOGGER.info("WebSocket opened: {}", session.getId()); sessions.add(session); } @OnMessage public void onMessage(ByteBuffer buffer, Session session) throws IOException { LOGGER.info("Received video data from {}", session.getId()); byte[] data = new byte[buffer.remaining()]; buffer.get(data); for (Session s : sessions) { if (s.isOpen() && !s.getId().equals(session.getId())) { s.getBasicRemote().sendBinary(ByteBuffer.wrap(data)); } } } @OnClose public void onClose(Session session) { LOGGER.info("WebSocket closed: {}", session.getId()); sessions.remove(session); } @OnError public void onError(Throwable error) { LOGGER.error("WebSocket error", error); } } ``` 6. 在前端页面中使用WebSocket实现视频通话。可以使用WebRTC等技术来实现视频采集、编解码、传输等功能。这里不再赘述。 以上就是一个简单的局域网音视频通话实现过程。需要注意的是,音视频通话涉及到的技术较多,需要根据实际情况进行选择和配置。

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